SYSTEMS, APPARATUSES AND METHODS FOR PARTY LINE CALLS AMONG VOICE OVER INTERNET PROTOCOL (VoIP) TELEPHONES

ABSTRACT

Devices, systems and methods are provided to support a party call with auto joining function in a communication system for a distributed transport system that employs VoIP technology. A VoIP unit or end point acts as a conference bridge between an operator extension and other end points. A first end point makes a SIP call to the operator and multicasts a mix of its send audio and the received operator audio. Other end points first check for multicast traffic being received before making a SIP call to the operator and, if so, play out the received audio over its speaker and multicast its send audio. The end point initiating the call mixes this subsequent stream into a stream to the operator, effectively acting as the conference bridge to the operator. The other end points are responsible for mixing received multicast streams for their local speaker.

This application claims the benefit of U.S. provisional application Ser.No. 62/469,620, filed Mar. 10, 2017, the entire content of which isincorporated herein by reference.

BACKGROUND OF THE INVENTION Field of the Invention

The present invention relates to systems, methods and apparatuses forVoIP telephones configured to be connected to an IP network (e.g., viafiber optic cables) and communicate with similarly configured VoIPtelephones to achieve a party fine function with automated joining of aparty line call.

Description of Related Art

Robust communications equipment and networks are often needed inenvironments subject to weather extremes and possible vandalism such asa railway system or subway system. For example, a railway system canhave a number of robust analog telephones installed at one or more trainstation(s) and along passenger platforms and in service or maintenanceareas and along the railway track (i.e., trackside). The telephones canbe relatively simple (e.g., a hands free configuration or handsetconfiguration) and have, for example, only one or two user buttons fordialing, For example, a hands free telephone can have a dedicated buttonor a handset telephone can “hot dial” upon lifting the handset to dialan operator (e.g., to report a condition or request assistance). Inelectrified transport networks (e.g., railway tracks, mining conveyorsystem, and so on), a telephone can also have a second, priority buttonfor calling the network's power control station to turn off power to thetracks or rails in an emergency situation.

Operators of transport networks can find a page party function to beuseful, that is, the ability for multiple telephones or user stations tobe involved on the same call. An approach that allows multipletelephones or user stations to be involved on the same call uses anetwork backbone as the communications infrastructure and Voice overInternet Protocol (VoIP) devices. With VoIP networks, multiple“channels” can be utilized over a single cable, it can he possible todeploy multiple IP devices across an existing network backbone, reducingthe cost of added cabling infrastructure significantly. Such traditionalVoIP solutions require the use of a VoIP exchange in much the samemanner that an analog telephony system would require an exchange.

Analog telephony systems, however, have many desirable functions thatare not available in traditional MAP systems, For example, in an analogsystem such as a residential system of parallel connected analogtelephones, a first user of a telephone in one location (e.g., in thekitchen) can he speaking to a called party, and second and third usersof other telephones (e.g., in the living room and in a bedroom) canautomatically join the call to the called party with the first user bymerely lifting the handset.

Such a multi-party auto joining function can be desirable in acommunication system for a distributed transport network or otherdistributed system that also employs VoIP technology and enjoys theadvantages of VoIP such as tower cost arid less cumbersome cabling. Aneed therefore exists for a VoIP unit or end point that can act as aconference bridle between an operator extension and other telephone endpoints. A need also exists for a VoIP solution for a communicationnetwork wherein multiple parties can automatically join a conferencecall or party call.

SUMMARY OF THE INVENTION

The above and other problems are overcome, and additional advantages arerealized, by illustrative embodiments of the present invention.

It is an aspect of illustrative embodiments of the present invention toprovide a voice over internet protocol (VoIP) unit that supports a partycall functionality with auto-joining comprising: a VoIP control modulefor VoIP operations among VoIP units in an Internet Protocol (IP)network, the VoIP control module configured to selectively establishpoint-to-point communication between its VoIP unit and another VoIPunit, and multicast communication among a plurality of the VoIP units;an IP network interface module connected to the VoIP control module andto the IP network and configured to exchange bi-directional Ethernetdata between the IP network and the VoIP control module; a speaker; amicrophone; and a user input device coupled to the VoIP control modulefor initiating a connection to at least one of an operator and one ormore of the other VoIP units via the IP network; wherein the VoIPcontrol module is further configured to initiate a call to an operatorand manage a party call with one or more of the other VoIP units, and tojoin a party call initiated by another one of the VoIP units.

In accordance with aspects of illustrative embodiments of the presentinvention, the VoIP unit, the operator, and the other VoIP units areregistered with a Session Initiation Protocol (SIP) server.

In accordance with aspects of illustrative embodiments of the presentinvention, when the VoIP unit initiates a call to the operator, the VoIPcontrol module is further configured to mix two audio streams togethercomprising the VoIP unit's send audio from the VoIP unit to the operatorand the operator's send audio from the operator to the VoIP unit, and toretransmit the two mixed audio streams as a multicast stream to theplurality of the VoIP units.

In accordance with aspects of illustrative embodiments of the presentinvention, wherein a second VoIP unit from among the plurality of theVoIP units determines if a multicast stream associated with the operatoris present before initiating a call to the operator. If a second VoIPunit determines that a multicast stream associated with the operator ispresent, then the second VoIP unit is configured to use the multicaststream associated with the operator as send audio from the operator andto transmit its send audio on a second multicast channel; and whereinthe VoIP control module of the VoIP unit that initiated the call to theoperator is configured to mix the second VoIP unit's send audio from thesecond multicast channel into the VoIP unit's multicast stream that isbroadcast to the other VoIP units and into an audio stream to theoperator. If the second VoIP unit determines that a multicast streamassociated with the operator is not present, then the second VoIP unitis configured to initiate a call to the operator.

In accordance with aspects of illustrative embodiments of the presentinvention, wherein a third VoIP unit determines that a multicast streamassociated with the operator is present and is configured to use themulticast stream associated with the operator as send audio from theoperator and to transmit its send audio on a third multicast channel;and wherein the VoIP control module of the VoIP unit that initiated thecall to the operator is configured to mix the third VoIP unit's sendaudio from the third multicast channel into the VoIP unit's multicaststream that is broadcast to the other VoIP units and into an audiostream to the operator,

In accordance with aspects of illustrative embodiments of the presentinvention, a method of automatically joining party line communicationamong plural voice over internet protocol (VoIP) units operating as endpoints in an IP network comprises registering plural VoIP units in an IPnetwork with a Session Initiation Protocol (SIP) server to supportpoint-to-point and multicast communications among them; a first VoIPunit initiating a call to a second VoIP unit selected from among theplural VoIP units, the first VoIP unit mixing two audio streams togethercomprising the first VoIP unit's send audio to the second VoIP unit andthe second VoIP unit's send audio to the first VoIP unit, andretransmitting the two mixed audio streams as a multicast stream on afirst multicast channel to the plurality of the VoIP units; and a thirdVoIP unit from among the plurality of the VoIP units determining if amulticast stream associated with the second VoIP unit is present beforeinitiating a call to the second VoIP unit. If the third VoIP unitdetermines that a multicast stream associated with the second VoIP unitis present, then the third VoIP unit uses the first multicast channelassociated with the second VoIP unit as send audio from the second VoIPunit and transmits the third VoIP unit's send audio on a secondmulticast channel. The first VoIP unit mixes the third VoIP unit's sendaudio from the second multicast channel into the first VoIP unit'smulticast stream that is broadcast to the other VoIP units.

In accordance with aspects of illustrative embodiments of the presentinvention, the second VoIP unit is an operator station and at least thefirst VoIP unit and the third VoIP unit each send their send audio tothe operator station via the first VoIP unit's multicast stream.

In accordance with aspects of illustrative embodiments of the presentinvention, if the third VoIP unit determines that a multicast streamassociated with the second VoIP unit is not present, then the third VoIPunit is configured to initiate a call to the operator.

Additional and/or other aspects and advantages of the present inventionwill he set forth in the description that follows, or will be apparentfrom the description, or may be learned by practice of the invention.The present invention may comprise VoIP units and systems and methodsfor operating same having one or more of the above aspects, and/or oneor more of the features and combinations thereof The present inventionmay comprise one or more of the features and/or combinations of theabove aspects as recited, for example, in the attached claims.

BRIEF DESCRIPTION OF THE DRAWINGS

The above and/or other aspects and advantages of embodiments of theinvention will be more readily appreciated from the following detaileddescription, taken in conjunction with the accompanying drawings, ofwhich:

FIG. 1 depicts an Internet. Protocol (IP) network in accordance with anembodiment of the present invention;

FIG. 2 is a flow chart depicting operations for party call functionalityamong VoIP units in accordance with an embodiment of the presentinvention;

FIGS. 3A and 3B are perspective views of illustrative housings for VoIPunits in accordance with embodiments of the present invention; and

FIG. 4 is a block diagram of components of a VoIP unit in accordancewith an embodiment of the present invention.

Throughout the drawing figures, like reference numbers will beunderstood to refer to like elements, features and structures.

DETAILED DESCRIPTION OF ILLUSTRATIVE EMBODIMENTS

Reference will now be made in detail to embodiments of the presentinvention, which are illustrated in the accompanying drawings. Theembodiments described herein exemplify, but do not limit, the presentinvention by referring to the drawings.

The present invention is described herein as a VoIP telephone configuredto be connected to an IP network (e.g., via fiber optic cables) andcommunicate with similarly configured VoIP telephones to achieve a partyline function with automated joining of a party call. A system of suchVoIP telephones is described in the context of a VoIP solution forreplacing existing trackside analog telephones in a railway system forillustrative purposes. It is to be understood, however, that the VoIPtelephone constructed in accordance with the present invention can beused in connection with different customers or applications besides atransportation system with rails or roadways or paths such as a networkof telephones in an industrial plant, power plant or manufacturingfacility, or mining system with conveyor, or multi-story parking garage,or park or golf course, for example. Further, the VoIP telephoneconstructed in accordance with the present invention can be hard wired(e.g., via fiber optic cables or copper) to an IP network.

Overview

Briefly, the party call or party line function can be accomplished viadifferent implementations. For example, with a conference bridgingmodel, a VoIP telephone (e.g., one of several optical tracksidetelephones in a railway system, or other end points) calls an operatorwhen this first trackside telephone initiates a conference. Other VoIPtelephones provided as trackside telephones are configured toautomatically join into the conference. A SIP server is used forregistration of all end points, that is, the trackside telephones andthe operator stations. Each end point registers with the SIP server andis able to setup or receive calls via the SIP server either one to one(e.g., point to point) or as a member of a group (e.g., multicast).Thus, when the operator calls a group, the SIP server renders theoperator group call into a call to each end point in that group.

Illustrative System 10

In accordance with an aspect of the present invention and as describedin more detail below in connection with FIG. 1, a plurality of opticaltrackside telephones (also hereinafter referred to as “end points” 16 or“VoIP units” 16) are deployed in string or ring topologies, with eachtelephone being linked only to its neighbors except for the two endunits or telephones 16 that are connected to a wider network via otherdevices. An example VoIP system 10 with party line functionality isdepicted in FIG. 1, wherein each of plural groups 12 a, . . . , 12 n oftelephones 16 has its own separate subnet and VLAN via fiber 14 a, . . ., 14 n to which ail group members 16 are connected (e.g., to constrainmulticast packets). The use of a VLAN allows local multicast to becontained within the group 12 or string. VLAN IDs need not be uniqueacross the system 10.

Due to the bidirectional fiber 14, switches 22 are configured for portblocking to prevent network loops. Each subnet terminates at a switch 22that supports a spanning tree protocol (e.g., STP) to prevent networkloops. Each end unit or telephone 16 in a group 12 can be connected to aswitch 22 ₁ and 22 ₂, that is, there can be two network switches perstring or group 12. Each switch 22 is connected to the wider network 18via a Layer 3 router 20, Since only one router is active at a time,routers 20 can run a redundancy protocol (e.g., virtual routerredundancy protocol or VRRP) so that a group 12 or string can survive aloss of one of the routers 20. The active router 20 operates as agateway for the string subnet of a group 12. Switches 22 1 and 22 ₂ in agroup 12 have direct connection between them (e.g., either a physicallink or on OSI Layer 2 connection through a service provider) via thenetwork cloud 18. In the event of a loss of a link within a string of agroup 12, a route remains from each telephone 16 to the active router 20or gateway and from each telephone 16 to all of the other telephones 16in the string (for multicast conferencing).

With continued reference to FIG. 1, a SIP server 24 and operator station26 are also connected to the cloud 13. Connections to the “cloud” ornetwork 18 are omitted from FIG. 1 for clarity. As stated above, thenetwork 18 consists of routers 20 forming a single Layer 3 networkarchitecture, supporting multiprotocol label switching (MPLS) or similardata-carrying protocol. To achieve redundancy, the VoIP telephones areconnected in a fiber ring 14, connected to the wider network 18 by meansof network routers 20. It is possible, however, to have a spur (i.e.,only one end terminates at a switch 22), resulting in a loss ofredundancy. The network 18 to which the fiber rings 14 of groups 12 areconnected is configured to allow large numbers of groups 12 to bedeployed, and with a group 12 having as many as 20 telephones 16, forexample, or more telephones. The network 18 allows telephones 16 in anygroup 12 to communicate with a common SIP server 24 in the networkwithout conflict. In addition, the groups 12 have sufficient resilienceto survive a loss of any one telephone 16 or link associated with eachgroup 12.

Illustrative Example of Party Line Functionality

The party line functionality among VoIP telephones is illustrated in theflowchart in FIG. 2, in accordance with an illustrative example, a firsttrackside device 16 makes a SIP call to the operator 26 (blocks 106 and108), and multicasts a stream comprising a mix of its send audio and thereceived operator audio (block 114). All of the other local end points(e.g., other trackside devices 16) are configured to be listening on therelevant port. If a second end point 16 goes off hook, it first checksto see if it is receiving multicast traffic before making a SIP call tothe operator 26 (block 106). If it is receiving multicast traffic (e.g.,the multicast from the first trackside device), then it plays out thereceived audio over its speaker and multicasts its send audio on aseparate multicast channel (block 110). The end point that initiated thecall (e.g., the first trackside device) mixes this second stream intoits Real-time Transport Protocol (RTP) stream to the operator (block112). Effectively, the initiating end point acts as the conferencebridge to the operator. All of the end points are responsible for mixingreceived multicast streams for their local speaker. The number ofsimultaneous multicast streams that can be mixed can be limited. Ifmulticast traffic associated with the intended SIP end point to becalled is not detected by an end point 16, then that end point wouldplace a SIP call to the operator 26.

More specifically, and with continued reference to FIG. 2, when a SIPcall is instigated between a telephone end point 16 a and an operatorend point 26 (block 108), RTP packets are sent bi-directionally betweenthe two end points carrying the two voice elements of the call, that is,the send audio from the telephone end point 16 a to the operator endpoint 26 and the send audio from the operator end point 26 to thetelephone end point 16 a. This first telephone end point I 6 a (e.g.,VoIP telephone constructed in accordance with illustrative embodimentsof the present invention) mixes the two audio streams together andre-transmits them as a multicast stream (block 114).

With continued reference to FIG. 2, before another telephone end point16 b places a call to the same operator extension 26, it checks for thepresence of the multicast stream associated with that operator extension26 (block 106). if it does not detect the presence of the requisitemulticast stream, then the other telephone end point 16 b places a SIPcall to the operator extension 26 (block 108). If it does detect thepresence of a requisite multicast stream, then the other telephone endpoint uses this multicast stream as the send audio from operator endpoint 26 and transmits its own send audio on a separate availablemulticast channel (block 110). The telephone end point 16 a thatoriginally instigated the SIP call to the operator extension 26 mixesthis second multicast stream and any subsequently received multicastaudio streams into its send RTP audio stream to the operator extension26 and into its broadcast audio multicast stream (block 112).

Subsequent telephone end points 16 n act in the same manner as end point16 h to join the party line call initiated by end point 16 a. If eitherthe instigating end point (e.g., first trackside device 16 a) oroperator 26 terminate the call, all instances terminate (blocks 116, 118and 122). If the call has not been terminated by the instigating endpoint or operator 26, but is terminated by any of the other telephoneend points 16 b, 16 n, then their respective instance terminates (block120). The call can also be terminated by an optional Call Limit Timerout. If the call has not been terminated by the instigating end point oroperator 26, but any of the other telephone end points 16 b, . . . , 16n has loss of multicast, then their respective instance terminates(block 122).

Illustrative VoIP Telephone 16

The user interface to a typical trackside fiber optic-connected VoIPhands free telephone 16 typically consists of a speaker 61 a, microphone61 b, an optional push to talk (PTT) non latching push button and, for atrackside telephone 16, a memory number auto dial /hook switchnon-latching push button(s) described below in a connection with FIGS.3A, SB and 4. Some VoIP units 16 may have up two separate memory numberauto dial non-latching push buttons (e.g., buttons 47 a and 47 b in FIG.3B). The microphone 61 b on the unit 16 will remain muted unless the PTTis depressed. On receiving a call, the alerting ring tone will beemitted from the speaker 52. The power requirements for these VoIPtelephones 16 can be fulfilled from batteries charged by a local solarpanel, for example. If a call is in progress, then pressing a memorynumber auto-dial push button will end the call; otherwise, it willinitiate a call to a preprogrammed number. Where a VoIP unit 16 has beenfitted with two memory number auto-dial push buttons 47 a, 47 b asillustrated in FIG. 3B, for example, pressing either push button willend a call that is in progress or initiate a call to the preprogrammednumber associated with that push button. Buttons can be optionallyprioritized. It can also be optional for a push button to end a call ifin its configuration. The number of fitted push buttons is only limitedby the number required for the application.

For hands free telephones 16 intended for public use such as at railroadlevel crossing, the power requirements can be fulfilled by a permanentsupply to the unit 16. Memory number auto dial push buttons can beconfigured to only instigate calls. Calls can he configured to only beterminated by the end party with the exception of a memory number autodial push button that has been configured as Priority. For example,where a unit 16 has been fitted with two memory number auto-dial pushbuttons, one of the push buttons can be configured as Priority suchthat, upon pressing the Priority push button, this will end anon-Priority call that is in progress and/or initiate a call to thepreprogrammed number associated with that push button.

Telephones connected on the same fiber segment 14 are assigned IPaddresses on the same local subnet. Each fiber segment 14 has a uniquelocal subnet. The number of trackside end points 16 on a given fibersegment 14 or local subnet can be limited (e.g., up to 20 end points 16,or other number of end points 16).

In the illustrated example of a VoIP system 10 such as a tracksidetelephone communication system, the vast majority of the tracksidetelephones 16 can be fitted with only a single memory number auto dialpush button, and the number associated with it will be for a controlroom operator 26 responsible for the particular section of railway wherethe telephone 16 is located.

Initiating a Call From the Trackside

With no call currently in progress, pressing a memory number auto dialpush button on a trackside telephone 16 instigates a SIP call to thenumber associated with the push button such as the number assigned to anoperator 26, The telephone of the called party (e.g., the operator 26)commences ringing and answers the incoming call, such as when theoperator 26 takes the telephone off hook. Two-way speech is establishedbetween the end points (i.e., the calling trackside telephone 16 and thecalled operator 26's station) but, if the optional PTT push button isfitted to the calling telephone 16, the microphone of the tracksidetelephone 16 is muted unless the PIT is depressed. The call isterminated when either party hangs up.

Initiating a Call From the Control Room

With no call currently in progress, the operator 26 can initiate a SIPcall to a specific end point 16. The end point 16 commences ringing andanswers the incoming call, such as when the user takes the telephone 16off hook (e.g., by pressing a push button, or removing a handset 61 froma cradle with a hook switch sensor). Only the end point 16 to which theoperator 26 made the call rings, Two-way speech is established betweenthe end point 16 and the operator 26 station but, if the optional PTTpush button is fitted to the trackside telephone 16, the microphone ofthe called trackside telephone 16 will he muted unless the P II isdepressed. The call is terminated when either party (e.g., the calledtelephone 16, or the operator 26) hangs up, lithe operator 26 needs tobe able to ring a number of trackside end points 16, then these numberswill be configured to be part of a Group 12 in the SIP exchange. Theoperator can then call the Group 12 n extension, which causes all of thetrackside end points 16 to ring at once. The first user to answer one ofthe ringing trackside telephones 16 is connected to the operator 26, andall of the remaining trackside telephones 16 cease to ring.

Party Call Functionality

If a memory number auto dial push button (e.g., button 47 a in FIG. 3B)is pressed on another unit 16 which is in the same local subnet of thegroup 12, then this end point automatically joins the alreadyin-progress call (i.e., party call) without intervention from either ofthe current users. There is no indication to the supplementary user(s)that a pre-existing call is in progress, The call persists until eitherthe operator or the trackside telephone 16 user that instigated the callhangs up. This is analogous with having the trackside telephonesconnected in parallel on an analog telephone line. For units 16 fittedwith a second memory number auto dial push button (e.g., button 47 b inFIG. 3B), any call initiated as a result of pressing the second pushbutton 47 b will be distinct from any other call in progress, althoughthe same party call functionality is available. Two, or other designatednumber of, different memory dial numbers are permitted on any localsubset.

An illustrative VoIP unit or end point 16 will now be described withreference to FIGS. 3A, 3B, and 4. The VoIP unit 16 is a low power, fiberoptic IP telephone such as a VoIP industrial telephone product availablefrom, for example, GAI-Tronics that uses an Analog Devices Inc. digitalsignal processor (DSP) and standard VoIP/SIP software. The VoIPindustrial telephone product has been customized to perform party lineoperations as described in connection with FIG. 2 in accordance with anaspect of an illustrative embodiment of the present invention.

FIGS. 3A and 3B depict, respectively, example end points 16 having ahandset 61 in an enclosure 63, and a no handset configuration (i.e., aspeaker 61 a and a microphone 61 b are integrated into the faceplate65). For example, FIG. 3A is shown with no dialer button(s) 47 a, 47 bsuch that filling the handset places a call on a predesignated “hotdial” memory number. In the application described above where end points16 are calling either an operator station 26 or control station, forexample, at least two memory number auto dial push buttons 47 a, 47 bcan be provided as shown in FIG. 3B. It is to be understood that otherkeypads can be provided to allow dialing to multiple stations or groups,and other telephone functions.

For example, the VoIP unit 16 can have up to 16 pushbutton inputs toenable a user to carry out actions such as starting a call. Buttonfunctions are defined and configurable in software. Configurablefunctions can include, but are not limited to, memory dial to apre-configured SIP URI, optional PTT, End call, Accept incoming call,and Toggle on off hook.

Pressing a button (e.g., button 47 a or 47 b) configured to make adirect call can cause the telephone 16 to initiate a call to a singleend point. The called end point 16 can either be a point-to-pointaddress or an extension on a SIP server 24. If a telephone 16 receivesan incoming one-to-one call, it can be set to either ring, in which casea designated button must be pressed to accept and answer, or toautomatically connect (with or without a preceding alert tone). Callsare ended either by pressing a “call end” button, by the receiving partyending the call, a configurable time out, or by the instigator of thecall, if the telephone has a PTT button, its microphone will be muted inany call unless this PTT button is held pressed. When in a call, thetelephone will be able to receive audio but not transmit unless the PTTbutton is pressed.

Example power and communication network cabling connections for anillustrative end point 16 such as a VoIP unit or telephone 16 will nowbe described. GAI-Tronics VoIP-SIP telephones, for example, supportconnection to a LAN or WAN. For example, the network connection can bevia two simplex LC connectors, each utilizing single mode fiber with abi-directional transceiver. One port can, for example, transmit at 1310nm and receive at 1550 nm, while the other port can transmit at 1550 nmand receive at 1310 nm. Network speed is typically 100 Mb/s. A USBconnector can also be provided on the printed circuit board (PCB) insidethe VoIP telephone 16 to enable a direct serial connection to, forexample, a laptop PC for configuration and management purposes. Theconnector is typically only accessible from inside the enclosure of theVoIP telephone 16.

Group 12 calls can be set up via a SIP server 24, but in this case thetelephone 16 treats the group identity as a SIP extension when making aone-to-many call. Alternatively, as described below, the party linefunctionality described with reference to FIG. 2 allows one or moretelephones 16 to automatically join a call instigated by anothertelephone 16 (e.g., such as a call to an operator station 26).

The server 26 is illustrated as a SIP server. By way of an example, SIPis used for messaging between end points 16 for creating, modifying andterminating point-to-point dialing, or multiparty (e.g., multicast)sessions consisting of one or several media streams. Multicast allows asingle audio stream to be received by multiple end points (e.g.,telephones 16) simultaneously to achieve multipoint paging or PublicAddress functionality over IP, as well as party line functionality inaccordance with an illustrative embodiment of the present invention.While each telephone 16 is configured or enabled to receive multicastpackets for multicast communications among themselves, amulticast-compliant SIP server 26 can be employed that supportsmulticast functionality such as a group call from an operator station 26to end points 16.

Each VoIP unit or telephone 16 can be configured with a dedicated SIPextension identifier (ID) and can have one or more multicast addressessuch that each telephone 16 can be a part of one or more groups.Accordingly, with its IP connectivity and loudspeaker, each telephone 16enables point-to-point calling with a “ring” tone generated to theloudspeaker, or multicast operation whereby the telephone 16 is able toreceive multicasts from other end points. The telephones 16 can then beconfigured to accept audio from one or more of these multicastaddresses, in addition to point to point calls to its dedicated SIPaddress.

With reference to FIG. 4, a VoIP unit or telephone 16 can be provided,for example, with a VoIP control board 44, a VoIP/phone interface board42, and a power board 40. The VoIP unit 16 is also provided with ahandset 61 with integrated microphone 61 b and speaker 61 a or handsfree microphone 61 b and speaker 61 a, and can also be a single boardimplementation. The VoIP control board 44 is described in more detailbelow. The VoIP/phone interface board 42 is connected to an Ethernetnetwork (e.g., a LAN or WAN to its group 12 and to the network 18 viaswitch 22) wirelessly or using fiber optic cable or copper, for example,via an Ethernet interface 14. The VoIP telephone 16 can be optionallyprovided with individual push buttons or a keypad board 46 whereapplicable, such as providing two or more call buttons 47 a, 47 b). TheVoIP unit or telephone 16 can have a power supply board 40 or, in analternative implementation, an amplifier and power supply board 40 foroperating a loudspeaker 52 to provide sufficiently loud audio outputsuitable for an industrial location that may have significant ambientnoise levels and/or inductive loop 51.

The VoIP unit or telephone 16 is provided with a magnetic hook switchsensor 60 (e.g., coupled to the handset 61's cradle), the output ofwhich can be coupled to the VoIP/phone interface board 42 for providingon-hook/off-hook status data 86 of the handset 61 to the VoIP controlboard 44. As stated above, the VoIP/phone interface board 42 isconfigured to provide public address (PA) speaker audio 68, as well asearpiece/speaker audio 62 from the VoIP control board 44 for the handsetor hands free speaker 61 a and receive microphone audio 64 from thehandset or separate microphone 61 b for the VoIP control hoard 44. TheVoIP/phone interface board 42 is also configured to provide individualpush buttons or keypad data from the keypad board 46 and DC power to the‘bolt’ control board 44. LEDs and relays are provided as generallyindicated at 54 for General Purpose Monitored Inputs/Outputs (I/O).

With continued reference to FIG. 4, power requirements for the VoIPtelephones 16 can be fulfilled from a battery/batteries charged by anexternal power source or a local solar panel, for example, Power supplyconditioning, the charging and condition monitoring of thebattery/batteries is performed by the power supply section 70, 71. Thebatteries are indicated in FIG. 4 as part of a power module 40 that canbe part of a single board implementation for a telephone 16. Audio 68for a speaker 52 is shown routed through the power module 40 to allowfor a public address (PA) function (e.g., where a larger amplifier isneeded). An audio induction loop indicated generally at 51 enables aperson using a hearing aid set to the ‘T’ setting to hear more clearly.

The VoIP/Phone interface board 42 and the VoIP control board 44 areconfigured to process Ethernet data 58. The VoIP control board 44 in atelephone 16 comprises a programmable processor 92 and memory 94. Inaccordance with illustrative aspects of the present invention, thetelephones 16 are programmed (e.g., via software code provided to theirrespective processors 92) to establish and terminate point-to-pointcalls and participate in party line calls, among other operations in theVoIP system 10 with party line functionality, As stated above, themicroprocessor 92 can be an Analog Devices Inc. ADSP-BF536 Seriesdigital signal processor (DSP) with standard VoIP/SIP software, forexample. The VoIP control board 44 employs an audio CODEC (e.g., 8 kHzG711A/U Law) to provide full duplex hands-free speech; that is, when ina call, the telephone 16's audio will be full duplex (i.e., transmit andreceive simultaneously with no switching).

In accordance with an aspect of an illustrative embodiment of thepresent invention, a VoIP industrial telephone product is customized(e.g., via software code processed by microprocessor 92) to performparty line operations as described in connection with FIG. 2. Inaddition, the VoIP control board 44 is programmed to support thefollowing protocols: Session Initiation Protocol (RFC3261), Payload forDTMF out-of-band digits (RFC2833), MIME Type Registration of RTP PayloadFormats (RFC3555), Simple Network Time Protocol (SNTP), Simple NetworkManagement Protocol (SNMP), Trivial File Transfer Protocol, SyslogProtocol (RFC3164/5424), Dynamic Host Control Protocol, Internet ControlMessage Protocol, User Datagram Protocol, and Internet Protocol IPv4 orIPv6.

As stated above, power requirements for the VoIP telephones 16 can befulfilled from batteries 70, 71 charged respectively by an externalpower source or a local solar panel. The power consumption of thetelephone 16 is kept to a minimum by keeping the processor core 92 andother hardware powered down or in sleep mode until it is actuallyneeded. In normal idle mode, only an onboard network switch can be fullyworking, that is, ready to react to any network packets destined for themain telephone 16 itself If the onboard network switch receives a packetnot intended for it, the packet will merely be forwarded to the secondport and not wake up the main processor 92. In this way the main, powerintensive components are switched off or in low power mode untiltriggered by an event such as: Receipt of an appropriate packet such asthe start of an incoming call or management access, Pressing a button,an alarm sensor such as low battery detection, or a timed activity suchas SIP re-registration or speaker-microphone testing.

The main method of configuring the VoIP unit telephone 16 is via SNMP,for example. Additionally it is possible to configure the telephone 16via a configuration file, which the telephone 16 can download viaTrivial File Transfer Protocol (TFTP). A Command Line Interface (CLI)interface is also provided, which is available over the Universal SerialBUS (USB) serial interface and Telnet protocol.

The telephones 16 are generally connected in a ring topology asillustrated in FIG. 1, 12 a . . . 12 n with two attributes. First, eachVoIP unit or telephone 16 controls network traffic passing through it soas to avoid packet storms, Second, should the ring be broken, forexample, by the loss of one VoIP unit 16 or connecting cable 14, thenetwork 12 a . . . 12 n will survive by switches 22 automaticallyunblocking a port to bypass the break. Recovery time from a ring break,that is, the time taken from the ring being broken to communicationbeing re-established with remaining accessible telephones 16, isgenerally no greater than 10 seconds. In practice, this is achieved byutilizing network switches 22 that support spanning tree protocols thatwill port block.

The following faults and alarms are configured for reporting, either viaSyslog or SNMP: Speaker / microphone fault, Low battery, Battery chargefailure (suppressed during darkness (night time), Stuck button, Powerinterruption, Registration failure, Configuration failure, Cold restart,Warm restart.

It will be understood by one skilled in the art that this disclosure isnot limited in its application to the details of construction and thearrangement of components set forth in the following description orillustrated in the drawings. The embodiments herein are capable of otherembodiments, and capable of being practiced or carried out in variousways, Also, it will be understood that the phraseology and terminologyused herein is for the purpose of description and should not be regardedas limiting. The use of “including,” “comprising,” or “having” andvariations thereof herein is meant to encompass the items listedthereafter and equivalents thereof as well as additional items. Unlesslimited otherwise, the terms “connected,” “coupled,” and “mounted,” andvariations thereof herein are used broadly and encompass direct andindirect connections, couplings, and mountings. In addition, the terms“connected” and “coupled” and variations thereof are not restricted tophysical or mechanical connections or couplings. Further, terms such asup, down, bottom, and top are relative, and are employed to aidillustration, but are not limiting.

The components of the illustrative devices, systems and methods employedin accordance with the illustrated embodiments of the present inventioncan be implemented, at least in part, in digital electronic circuitry,analog electronic circuitry, or in computer hardware, firmware,software, or in combinations of them. These components can beimplemented, for example, as a computer program product such as acomputer program, program code or computer instructions tangiblyembodied in an information carrier, or in a machine-readable storagedevice, for execution by, or to control the operation of, dataprocessing apparatus such as a programmable processor, a computer, ormultiple computers. A computer program can he written in any form ofprogramming language, including compiled or interpreted languages, andit can be deployed in any form, including as a stand-alone program or asa module, component, subroutine, or other unit suitable for use in acomputing environment. A computer program can be deployed to be executedon one computer or on multiple computers at one site or distributedacross multiple sites and interconnected by a communication network.Also, functional programs, codes, and code segments for accomplishingthe present invention can be easily construed as within the scope of theinvention by programmers skilled in the art to which the presentinvention pertains, Method steps associated with the illustrativeembodiments of the present invention can be performed by one or moreprogrammable processors executing a computer program, code orinstructions to perform functions (e.g., by operating on input dataand/or generating an output). Method steps can also be performed by, andapparatus of the invention can be implemented as, special purpose logiccircuitry, e.g., an FPGA (field programmable gate array) or an ASIC(application-specific integrated circuit).

Processors suitable for the execution of a computer program include, byway of example, both general and special purpose microprocessors, andany one or more processors of any kind of digital computer. Generally, aprocessor will receive instructions and data from a read-only memory ora random access memory or both, The essential elements of a computer area processor for executing instructions and one or more memory devicesfor storing instructions and data. Generally, a computer will alsoinclude, or he operatively coupled to receive data from or transfer datato, or both, one or more mass storage devices for storing data, e.g.,magnetic, magneto-optical disks, or optical disks. Information carrierssuitable for embodying computer program instructions and data includeall forms of non-volatile memory, including by way of example,semiconductor memory devices, e.g. EPROM, EEPROM, and flash memorydevices; magnetic disks, e.g., internal hard disks or removable disks;magneto-optical disks; and CD-ROM and DVD-ROM disks. The processor andthe memory can be supplemented by, or incorporated in special purposelogic circuitry.

Those of skill in the art would understand that information and signalsmay be represented using any of a variety of different technologies andtechniques. For example, data, instructions, commands, information,signals, bits, symbols, and chips that may be referenced throughout theabove description may be represented by voltages, currents,electromagnetic waves, magnetic fields or particles, optical fields orparticles, or any combination thereof.

Those of skill in the art would further appreciate that the variousillustrative logical blocks, modules, circuits, and algorithm stepsdescribed in connection with the embodiments disclosed herein may beimplemented as electronic hardware, computer software, or combinationsof both. To clearly illustrate this interchangeability of hardware andsoftware, various illustrative components, blocks, modules, circuits,and steps have been described above generally in terms of theirfunctionality. Whether such functionality is implemented as hardware orsoftware depends upon the particular application and design constraintsimposed on the overall system, Skilled artisans may implement thedescribed functionality in varying ways for each particular application,but such implementation decisions should not be interpreted as causing adeparture from the scope of the present invention.

The various illustrative logical blocks, modules, and circuits describedin connection with the embodiments disclosed herein may be implementedor performed with a general purpose processor, a Digital SignalProcessor (DSP), an Application Specific Integrated Circuit (ASIC), aField Programmable Gate Array (FPGA) or other programmable logic device,discrete gate or transistor logic, discrete hardware components, or anycombination thereof designed to perform the functions described herein,A general purpose processor may be a microprocessor, but in thealternative, the processor may be any conventional processor,controller, microcontroller, or state machine. A processor may also beimplemented as a combination of computing devices, e.g., a combinationof a DSP and a microprocessor, a plurality of microprocessors, one ormore microprocessors in conjunction with a DSP core, or any other suchconfiguration.

The steps of a method or algorithm described in connection with theembodiments disclosed herein may be embodied directly in hardware, in asoftware module executed by a processor, or in a combination of the two.A software module may reside in Random Access Memory (RAM), flashmemory, Read Only Memory (ROM), Electrically Programmable ROM (EPROM),Electrically Erasable Programmable ROM (EEPROM), registers, hard disk, aremovable disk, a CD-ROM, or any other form of storage medium known inthe art. An exemplary storage medium is coupled to the processor suchthe processor can read information from, and write information to, thestorage medium. In the alternative, the storage medium may be integralto the processor. The processor and the storage medium may reside in anASIC. The ASIC may reside in the remote station, Electronic medicaldevice, a server, or a combination thereof. In the alternative, theprocessor and the storage medium may reside as discrete components in auser terminal.

The previous description of the disclosed embodiments is provided toenable any person skilled in the art to make or use the presentinvention, Various modifications to these embodiments will be readilyapparent to those skilled in the art, and the generic principles definedherein may be applied to other embodiments without departing from thespirit or scope of the invention. Thus, the present invention is notintended to be limited to the embodiments shown herein but is to beaccorded the widest scope consistent with the principles and novelfeatures disclosed herein.

The above-presented description and figures are intended by way ofexample only and are not intended to limit the present invention in anyway except as set forth in the following claims. It is particularlynoted that persons skilled in the art can readily combine the varioustechnical aspects of the various elements of the various illustrativeembodiments that have been described above in numerous other ways, allof which are considered to he within the scope of the invention.

1. A voice over internet protocol (VoIP) unit comprising: a VoIP controlmodule for VoIP operations among VoIP units in an Internet Protocol (IP)network, the VoIP control module configured to selectively establishpoint-to-point communication between its VoIP unit and another VoIPunit, and multicast communication among a plurality of the VoIP units;an IP network interface module connected to the VoIP control module andto the IP network and configured to exchange hi-directional Ethernetdata between the IP network and the VoIP control module; a speaker; amicrophone; and a user input device coupled to the VoIP control modulefor initiating a connection to at least one of an operator and one ormore of the other VoIP units via the IP network; wherein the VoIPcontrol module is further configured to initiate a call to an operatorand manage a party call with one or more of the other VoIP units, and tojoin a party call initiated by another one of the VoIP units.
 2. A VoIPunit as recited in claim 1, wherein the VoIP unit, the operator, and theother VoIP units are registered with a Session Initiation Protocol (SIP)server.
 3. A VoIP unit as recited in claim 1, wherein, when the VoIPunit initiates a call to the operator, the VoIP control module isfurther configured to mix two audio streams together comprising the VoIPunit's send audio from the VoIP unit to the operator and the operator'ssend audio from the operator to the VoIP unit, and to retransmit the twomixed audio streams as a multicast stream to the plurality of the VoIPunits.
 4. A VoIP unit as recited in claim 3, wherein a second VoIP unitfrom among the plurality of the VoIP units determines if a multicaststream associated with the operator is present before initiating a callto the operator.
 5. A VoIP unit as recited in claim 3, wherein, if asecond VoIP unit determines that a multicast stream associated with theoperator is present, then the second Vole unit is configured to use themulticast stream associated with the operator as send audio from theoperator and to transmit its send audio on a second multicast channel;and wherein the VoIP control module of the VoIP unit that initiated thecall to the operator is configured to mix the second VoIP unit's sendaudio from the second multicast channel into the VoIP unit's multicaststream that is broadcast to the other VoIP units and into an audiostream to the operator.
 6. A VoIP unit as recited in claim 5, wherein,if the second VoIP unit determines that a multicast stream associatedwith the operator is not present, then the second VoIP unit isconfigured to initiate a call to the operator.
 7. A VoIP unit as recitedin claim 5, wherein a third VoIP unit determines that a multicast streamassociated with the operator is present and is configured to use themulticast stream associated with the operator as send audio from theoperator and to transmit its send audio on a third multicast channel;and wherein the VoIP control module of the VoIP unit that initiated thecall to the operator is configured to mix the third VoIP unit's sendaudio from the third multicast channel into the VoIP multicast streamthat is broadcast to the other VoIP units and into an audio stream tothe operator.
 8. A VoIP unit as recited in claim 1, wherein, if the VoIPunit determines that a multicast stream associated with the operator ispresent, then the VoIP unit is configured to use the multicast streamassociated with the operator as send audio from the operator and totransmit its send audio on a second multicast channel; and if the secondVoIP unit determines that a multicast stream associated with theoperator is not present, then the VoIP unit is configured to initiate acall to the operator.
 9. A method of automatically joining party linecommunication among plural voice over internet protocol (VoIP) unitsoperating as end points in an IP network comprising: registering pluralVoIP units in an IP network with a Session Initiation Protocol (SIP)server to support point-to-point and multicast communications amongthem; a first VoIP unit initiating a call to a second VoIP unit selectedfrom among the plural VoIP units, the first VoIP unit mixing two audiostreams together comprising the first VoIP unit's send audio to thesecond VoIP unit and the second VoIP unit's send audio to the first VoIPunit, and retransmitting the two mixed audio streams as a multicaststream on a first multicast channel to the plurality of the VoIP units;a third VoIP unit from among the plurality of the VoIP units determiningif a multicast stream associated with the second VoIP unit is presentbefore initiating a call to the second VoIP unit; if the third VoIP unitdetermines that a multicast stream associated with the second VoIP unitis present, then the third VoIP unit uses the first multicast channelassociated with the second VoIP unit as send audio from the second VoIPunit and transmits the third VoIP unit's send audio on a secondmulticast channel; the first VoIP unit mixing the third VoIP unit's sendaudio from the second multicast channel into the first VoIP unit'smulticast stream that is broadcast to the other VoIP units
 10. A methodof automatically joining party line communication among plural VoIPunits as recited in claim 9, wherein the second VoIP unit is an operatorstation and at least the first VoIP unit and the third VoIP unit eachsend their send audio to the operator station via the first VoIP unit'smulticast stream.
 11. A method of automatically joining party linecommunication among plural VoIP units as recited in claim 9, wherein, ifthe third VoIP unit determines that a multicast stream associated withthe second VoIP unit is not present, then the third VoIP unit isconfigured to initiate a call to the operator.